Wide area dispatch systems are well known in the art for providing simultaneous communications to a large number of mobile, geographically distributed users. Prior art wide area systems are typically composed of multiple geographically distributed radio sites connected to a master location, via point-to-point dedicated telecommunications circuits. At call initiation, or setup, a dedicated communications path is assigned to each radio site that is to participate in the call (i.e., sites whose coverage area services the target communication units). Further, a dedicated voice bridge, located at the master location, is assigned to the call. Upon receipt of the inbound audio from the initiating communication unit, the receiving radio site transfers the voice on to the dedicated communications circuit. The central site presents the voice to the voice bridge, which in turn produces multiple copies of the voice, one for each target or receiving radio site.
The foregoing system, while perhaps generally solving the problem of dispatch audio, is inherently inefficient due to the dedication of a communications circuit and a voice bridge for each call. That is, a significant portion of the communications infrastructure remains idle while a call is being assigned. As an example, while the speaker pauses during the call (e.g., as between two sentences or thoughts), the communications facilities remain dedicated to the call, while no information is being transferred. In particular, the expensive communications facilities (e.g., a dedicated 64 kbps circuit) is not shared among multiple users while the facility is assigned to a call, since the switching infrastructure differentiates calls on the basis of assigned communications facilities. Accordingly, this results in the systematic duplication of costly communications infrastructure to support a greater number of calls.
One attempt to solve the problem has been to digitize the voice into discrete packets, and switch the discrete packets using an X.25 packet switch. However, this solution is feasible only for systems that deliver data only (e.g., to/from computers, printers and routers), and not voice. That is, these devices typically require that the information be received error free, while tolerating rather large delays (e.g., in the range of seconds) as well as significant variance between delays among sequential packets (e.g., one packet received 500 milliseconds late, and the next packet received 100 milliseconds late). In particular, an X.25 switch, upon detection of a corrupted message, typically discards the entire message and requests that a second copy of the message be sent, thus preserving the integrity of the message at the expense of additional delay. (It should be noted that, statistically speaking, not all messages are corrupted. Thus, the observed message delivery delay varies depending on the number of transmissions required to successfully deliver the message to the intended destination.) Of course, in a voice-capable system, it would be impractical to require the speaker to re-send the corrupted voice message. Likewise, it would be unacceptable to have delays of this order in a voice message.
Voice, in stark contrast to data, imposes different requirements upon the transmission facilities. That is, an ideal transmission of a voice message would forego message integrity to provide tighter delay parameters. Unlike a data-oriented computer system, humans are well adapted at "filling-in" any missing gaps of audio. That is, the listener need not hear every syllable of every word in a message correctly to derive the information intended to be communicated by the speaker. It is only important that the syllables of the message be timely delivered in a consistent manner, so the listener can "fill-in" the gaps formed when one of the syllables is incorrectly transmitted (i.e., after modulation). In short, a voice message may be usable if delivered with a small percentage of the information lost and slightly delayed, but is far less usable when delivered with a significant variance in the message delivery delay. That is, the integrity of voice--while preserved in terms of content--is destroyed by the high statistical inter-message delay associated with the X.25 communications facility. Furthermore, an X.25 switch does not provide the ability for dynamic reconfiguration of services that are needed to replace the voice bridge. Likewise, conventional frame relay is inadequate, as it fails to provide a platfom, for reliable, concurrent transmission of voice and data (e.g., control data).
Accordingly, a need exists for a method and an apparatus to establish audio communication between communication units in a wide area system that is not constrained by the shortcomings found in the prior art. In particular, a voice communication system that minimizes the magnitude and variance of inter-message delay, while making efficient use of the costly infrastructure, would be an improvement over the prior art.